Readers,

I am having issues with passing the max-forward header to my freeswitch service.

Here are the sip invites 

Here is a call coming from a kazoo box though my carrier Into kamailio. Kamailio shows it has Max-Forward after the sanitize check, but when it reaches freeswitch no Max-Forward Header.

KAMAILIO

kamailio[2687]: ERROR: <script>: After Sanitize - [INVITE sip:+1850764####@aio07-bandwidth.tresta-aio.com:5060 SIP/2.0
Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8>
Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8>
Accept: application/sdp
Allow: INVITE,ACK,CANCEL,BYE
Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.2791522.0
Via: SIP/2.0/UDP 67.231.8.###;branch=z9hG4bKc892.f8621604.0
Via: SIP/2.0/UDP 67.231.9.###:5060;branch=z9hG4bK04B40f39b0e10295cd4
From:  <sip:+1NUMBERSCRUBED@67.231.9.59;isup-oli=0>;tag=gK043cdaa8
To: <sip:+1850764####@67.231.8.85>
Call-ID: 537168748_70701748@67.231.9.###
CSeq: 1289511105 INVITE
Max-Forwards: 49
Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060>
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
X-FS-Support: update_display,send_info
Supported: precondition
Content-Length:   359
Content-Type: application/sdp
Remote-Party-ID:  <sip:+1NUMBERSCRUBED@67.231.9.###:5060>;privacy=off;screen=no

v=0
o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59
s=SIP Media Capabilities
c=IN IP4 67.231.9.72
t=0 0
m=audio 23918 RTP/AVP 9 0 18 96 101
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=maxptime:30

FREESWITCH

recv 1479 bytes from udp/[10.1.13.123]:5060 at 18:31:26.312854:
   ------------------------------------------------------------------------
   INVITE sip:+1850764###@aio07-bandwidth.tresta-aio.com:5060 SIP/2.0
   Record-Route: <sip:10.1.13.###;lr=on;ftag=gK043cdaa8>
   Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8>
   Record-Route: <sip:67.231.8.###;lr=on;ftag=gK043cdaa8>
   Accept: application/sdp
   Allow: INVITE,ACK,CANCEL,BYE
   Via: SIP/2.0/UDP 10.1.13.123;branch=z9hG4bKc892.1b308e66.0
   Via: SIP/2.0/UDP 67.231.8.195;branch=z9hG4bKc892.2791522.0
   Via: SIP/2.0/UDP 67.231.8.85;branch=z9hG4bKc892.f8621604.0
   Via: SIP/2.0/UDP 67.231.9.59:5060;branch=z9hG4bK04B40f39b0e10295cd4
   From:  <sip:+1NUMBERSCRUBED@67.231.9.###;isup-oli=0>;tag=gK043cdaa8
   To: <sip:+1850764####@67.231.8.###>
   Call-ID: 537168748_70701748@67.231.9.###
   CSeq: 1289511105 INVITE
   MaxForwards would go here?
   Contact: "1NUMBERSCRUBED" <sip:+1NUMBERSCRUBED@67.231.9.###:5060>
   Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
   Supported: precondition
   Content-Length:   359
   Content-Type: application/sdp
   Remote-Party-ID:  <sip:+1NUMBERSCRUBED@67.231.9.###:5060>;privacy=off;screen=no
   X-AUTH-IP: 67.231.8.195

   v=0
   o=Sonus_UAC 1180063305 112694125 IN IP4 67.231.9.59
   s=SIP Media Capabilities
   c=IN IP4 67.231.9.72
   t=0 0
   m=audio 23918 RTP/AVP 9 0 18 96 101
   a=rtpmap:9 G722/8000
   a=rtpmap:0 PCMU/8000
   a=rtpmap:18 G729/8000
   a=fmtp:18 annexb=no
   a=rtpmap:96 iLBC/8000
   a=fmtp:96 mode=30
   a=rtpmap:101 telephone-event/8000
   a=fmtp:101 0-15
   a=sendrecv
   a=maxptime:30

Here is my kamailio default.cfg http://pastebin.com/vxdFe8n0

Can anyone point me in the right direction, and tell me why freeswitch isn't being passed this header? Both kamailio and freeswitch are on the same box in this example.

Thanks for your help.
Mike